6 Commits

Author SHA1 Message Date
solenberg
3ebbcb528b Stop using VoEVideoSync in Call/VideoReceiveStream.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2452163004
Cr-Commit-Position: refs/heads/master@{#16375}
2017-01-31 11:58:40 +00:00
asapersson
fe50b4d750 Make class of static functions in rtp_to_ntp.h:
- UpdateRtcpList
- RtpToNtp

class RtpToNtpEstimator
- UpdateMeasurements
- Estimate

List with rtcp measurements is now private.

BUG=none

Review-Url: https://codereview.webrtc.org/2574133003
Cr-Commit-Position: refs/heads/master@{#15762}
2016-12-22 15:53:51 +00:00
asapersson
b7e7b49551 Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
BUG=webrtc:6579

Review-Url: https://codereview.webrtc.org/2435053004
Cr-Commit-Position: refs/heads/master@{#15125}
2016-11-17 10:27:20 +00:00
asapersson
de9e5fffa2 Add stats for frequency offset when converting RTP timestamp to NTP time.
- Add histogram: "WebRTC.Video.RtpToNtpFreqOffsetInKhz"

  The absolute value of the difference between the estimated frequency during RTP timestamp to NTP time conversion and the actual value (i.e. 90 kHz) is measured per received video frame. The max offset during 40 second intervals is stored. The average of these stored offsets per received video stream is recorded when a stream is removed.

Updated rtp_to_ntp.cc:
- Add validation for only inserting newer RTCP sender reports to the rtcp list.

- Move calculation of frequency/offset (from RTP/NTP timestamp pairs) to UpdateRtcpList. Calculated when a new RTCP SR in inserted (and not in RtpToNtpMs per packet).

BUG=webrtc:6579

Review-Url: https://codereview.webrtc.org/2385763002
Cr-Commit-Position: refs/heads/master@{#14891}
2016-11-02 14:14:10 +00:00
asapersson
b0c1b4e24d Do not update stream synchronization if no new video packet has been received since last update (e.g. video muted).
BUG=

Review-Url: https://codereview.webrtc.org/2334113004
Cr-Commit-Position: refs/heads/master@{#14271}
2016-09-17 08:00:04 +00:00
mflodman
4cd2790f17 Move RTP for synchroninzation and rename classes, files and variables.
This CL removes (almost) the last RTP references in VideoReceiveStream.
There are still references to RTPFragmentationHeader and SSRCs, which
will be dealt with later.

There are also new GUARDED_BY and thred checker added to the
synchronization class.

When there are othre transports than RTP, there will instead be an
interface + inheritance for RtpStreamReceiver and
RtpStreamSynchronizattion in VideoReceiveStream. This work will be done
when we actually know how we want to make thee transport interface.

BUG=webrtc:5838

Review-Url: https://codereview.webrtc.org/2216533002
Cr-Commit-Position: refs/heads/master@{#13655}
2016-08-05 13:28:50 +00:00